similar to: Obi110 as gateway to PSTN?

Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as "sip:user at domain.com", such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hello I successfully hooked up a Linksys 3102 SIP gateway (http://www.voip-info.org/users/683/21683/images/716/SPA3102_lrg.jpg) to an Asterisk server, but since it's connected to a PSTN line, I must make sure it cannot be used by unauthorized users from the Net. Actually, even legit users with an account on the Asterisk server shouldn't be able to use it (outgoing calls should go

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

Hello To run Asterisk on an embedded appliance, ie. where RAM and non-volatile memory is an issue (respectively 64MB and 256MB), I need to check how much space voice messages take to save and play back. The appliance is connected to a landline in Europe (in case that makes a difference as far as codecs are concerned). Is there a document that shows the different options, with/without

HI, I'd like to redirect an incoming E1 call to a local landline, at the moment I just do Exten => thenumber,1,Dial(Zap/g1/localnumber) However this seems to cause all sorts of problems with the fax machine on the end of that landline. Is there a better way to redirect a call? Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.

Hello Gurus I've installed my Asterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I

Putting the Genie Back in the Bottle: More RedHat Legal Shenanigans with CentOS http://nerdvittles.com/?p=8721 Closing the Book on CentOS: [...] http://nerdvittles.com/ Thanks to the author of the above articles. jb

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually unlimited Khz mono files (still no stereo, but a quantum leap forward). I've played with this and get good throughputs using SLIN44 formats on SIP. The 2 questions

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part

Yo all, The PSTN access-number for the Dutch IAXTel <-> PSTN-gateway has changed. The new number is: +31 20 3987 567. Calling from IAXTel to Dutch toll-free PSTN-numbers is still done in the same way, by calling "31800<rest of number>". Mark: Could you please update your web-sites to reflect this change? The old number is mentioned on "http://www.gnophone.com/",

Hello list, I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is connected to an * server and i have 10 users using this setup. I do have some problems in establishing a call to an outside location (call that goes through the SPA400). The first attempt doesn't get through. I suspect the spa400 being the source of the problem. The Linksys SPA400 has a lot of params on the

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp:

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to PSTN network (carrier). I need to configure * to